[..] |
ASTERISK-26000: testsuite: Create pilot test for ARI channels/create and channels/dial |
ASTERISK-26001: cel, odbc: Asterisk ignores dateformat setting in cel.conf and default behaviour wrong |
ASTERISK-26002: CEL: Add dialed extension for attended transfers to non-phone extensions |
ASTERISK-26003: Unexpected call termination when using PJSIP + SRTP |
ASTERISK-26004: res_pjsip: The transport/method parameter is ignored |
ASTERISK-26005: res_pjsip: Multiple SIP messages are combined into 1 TCP packet |
ASTERISK-26006: Show offending IP for TLS setup failures in logs |
ASTERISK-26007: res_pjsip: Endpoints deleting early after upgrade from 13.8.2 to 13.9 |
ASTERISK-26008: app_followme does not delete recorded name prompt |
ASTERISK-26009: res_ari: Asterisk crashes with several ARI apps connected with subscribeAll=1 |
ASTERISK-26010: [patch]func_odbc: single database connection should be optional |
ASTERISK-26011: [patch]PJSIP: add "via_addr", "via_port", "call_id" to contacts |
ASTERISK-26012: How can I put the call on Hold/Unhold and its event |
ASTERISK-26013: Multiple queued calls sent to agent |
ASTERISK-26014: res_sorcery_astdb: Make tolerant of unknown fields |
ASTERISK-26015: Registering a SIP Account using CLI/PHP |
ASTERISK-26016: Dialplan execution error in h ext |
ASTERISK-26017: dial: Crash when executing pre-dial handler |
ASTERISK-26018: res_ari: Empty bridge displayed even after destruction |
ASTERISK-26019: Deadlock after 'core reload' |
ASTERISK-26020: memory leak |
ASTERISK-26021: Build codecs siren7 and siren14 for Asterisk 13 |
ASTERISK-26022: ARI: Add media playlists |
ASTERISK-26023: app_mixmonitor: Sound distortion while "End MixMonitor Recording SIP/XXXX" |
ASTERISK-26024: 13.8 Ami response to channel status action is incomplete |
ASTERISK-26025: core: Asterisk doesn't start correctly in kickstart post-install script |
ASTERISK-26026: res_pjsip: Error on configuring PJSIP multi domain |
ASTERISK-26027: res_pjsip_exten_state: Testsuite tests for extension state publishing |
ASTERISK-26028: WebRTC one sec delay hearing sound |
ASTERISK-26029: parking: ast_parking_park_call should return parking_space instead of parking_exten |
ASTERISK-26030: call cut because of double Session-Expires header in re-invite after proxy authentication is required |
ASTERISK-26031: contrib: RHEL init script does not show "OK" or "ERROR" on start |
ASTERISK-26032: Pause with reason "wrap-up" for after-call-work |
ASTERISK-26033: PJSIP unable to register or qualify to IMS Trunk using outbound proxy |
ASTERISK-26034: T.38 passthrough problem behind firewall due to early nosignal packet |
ASTERISK-26035: PJSIP driver, TLS client in ringgroup crashes asterisk with incoming trunk call |
ASTERISK-26036: res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk |
ASTERISK-26037: Asterisk crashed repeatedly - Segmentation fault |
ASTERISK-26038: 'make install' doesn't seem to install OS/X init files |
ASTERISK-26039: chan_sip: Failure to INVITE authentication. (When the From header is "Anonymous".) |
ASTERISK-26040: Outbound Caller ID Issue after E1 Trunking |
ASTERISK-26041: odbc: Asterisk crashed repeatedly - ODBC connector |
ASTERISK-26042: ARI: Allow downloading of the media associated with a stored recording |
ASTERISK-26043: core: Crash when notifying threadpool listener |
ASTERISK-26044: chan_sip: Realtime peer matching wrong peer despite match_auth_username=yes |
ASTERISK-26045: [patch]app_voicemail: fix bugs, imap mm_status log change to debug |
ASTERISK-26046: [patch] Avoid obsolete warnings on autoconf. |
ASTERISK-26047: ARI allows certain commands to run on down channels. |
ASTERISK-26048: Asterisk crashes with PJSIP Assertion "Invalid transport name" |
ASTERISK-26049: res_pjsip: Crash when our own request timer fires |
ASTERISK-26050: chan_sip: WebRTC Audio + Video Negotiation Problem |
ASTERISK-26051: Audiocodes MP-114FXO with Asterisk PBX configuration |
ASTERISK-26052: Asterisk hangs with a T.38 call |
ASTERISK-26053: res_pjsip_outbound_publish: Crash when shutting down |
ASTERISK-26054: Asterisk crashes (core dump) |
ASTERISK-26055: [patch]res_pjsip: chatty verbose messages |
ASTERISK-26056: res_config_mysql complains about Alembic MySQL schema |
ASTERISK-26057: res_config_sqlite3 uses incorrect query - unnecessary escape |
ASTERISK-26058: [Patch] Add uptime and last reloaded to FullyBooted AMI event |
ASTERISK-26059: [patch]core: New channel variable FORWARDERNAME |
ASTERISK-26060: Asterisk Dead Lock |
ASTERISK-26061: [patch] res_pjsip: improve realtime performance - remove updating all endpoints status on startup |
ASTERISK-26062: Testsuite: tests/bridge/dial_LS_options sporadically fails |
ASTERISK-26063: ${PJSIP_HEADER(read,Call-ID)} does not work - documentation needs clarification for when read/write is possible |
ASTERISK-26064: followme: allow disabling callee prompt |
ASTERISK-26065: chan_pjsip: MWI NOTIFY contents not ordered properly |
ASTERISK-26066: Error type the SIP username= "375249700566@ims.alter.cu" |
ASTERISK-26067: chan_sip: Asterisk dialplan hangs on a 423 |
ASTERISK-26068: Multicast RTP Options |
ASTERISK-26069: Asterisk truncates To: header, dropping the closing '>' |
ASTERISK-26070: ari/channels: Creating a local channel without an originator adds all audio formats to it's capabilities |
ASTERISK-26071: SIP peer poke info has to redirect to my java application |
ASTERISK-26072: chan_sip: Asterisk 13 don't send INVITE with Replaces after accepted REFER |
ASTERISK-26073: chan_sip: Crash when cleaning up pruned peers |
ASTERISK-26074: res_odbc: Deadlock within UnixODBC |
ASTERISK-26075: Impossible to define two TCP or TLS transports with pjsip |
ASTERISK-26076: Show user agent in PJSIPShowEndpoint |
ASTERISK-26077: safe_asterisk ulimit can be above kernel max |
ASTERISK-26078: core: Memory leak in logging |
ASTERISK-26079: Regular Segfaults of Asterisk w/ARI |
ASTERISK-26080: One-way audio on incoming chan_mobile calls |
ASTERISK-26081: I need to customize calling permission like -intercom, local, national, international, etc |
ASTERISK-26082: res_pjsip_messaging: MessageSend Content-Type can't be changed |
ASTERISK-26083: ARI: Announcer channels staying around after playback to a bridge is finished |
ASTERISK-26084: Support wideband audio for app_mp3 |
ASTERISK-26085: app_mp3: results in timeout for streams |
ASTERISK-26086: res_musiconhold: format option is not documented adequately |
ASTERISK-26087: Icelandic grammar support for voicemail and numbers |
ASTERISK-26088: Investigate heavy memory utilization by res_pjsip_pubsub |
ASTERISK-26089: Invalid security events during boot using PJSIP Realtime |
ASTERISK-26090: res_http_websocket: Crash when reading message |
ASTERISK-26091: [patch] ar cru creates warning, instead use ar cr |
ASTERISK-26092: [Segfault] in res_rtp_asterisk.c:4268 after Remotely bridged channels |
ASTERISK-26093: Program terminated with signal 11, Segmentation fault. |
ASTERISK-26094: stasis: Playing MOH to bridge with ARI does not work |
ASTERISK-26095: chan_iax2: Deadlock |
ASTERISK-26096: res_hep: Crash when configuration file is missing |
ASTERISK-26097: [patch] CLI: show maximum file descriptors |
ASTERISK-26098: cdr: Source and Destination on the middle leg of a blonde transfer are incorrect |
ASTERISK-26099: res_pjsip_pubsub: Crash when sending request due to server timeout |
ASTERISK-26100: [patch] Asterisk will reject, with a 488, subsequent T.38 reinvite to T.38 negotiation. |
ASTERISK-26101: [patch] Bridge: Channel is not unheld in attended transfer |
ASTERISK-26102: Manager Event DTMF not generated if call comes from a queue |
ASTERISK-26103: cdr: Assert on 'dial end' event during a blond transfer |
ASTERISK-26104: Build: A Fedora GLIBC update to 2.22-17 causes a compile failure |
ASTERISK-26105: pjsip: PJSIP_HEADER doesn't add SIP header |
ASTERISK-26106: sorcery: PJSIP driver completely broken with oracle using ODBC |
ASTERISK-26107: chan_sip: ASAN: heap-buffer-overflow on sip reload |
ASTERISK-26108: parking: Cannot create parking lots with extension over 2147483647 |
ASTERISK-26109: Asterisk fails building with OpenSSL 1.1.0 |
ASTERISK-26110: bridge: Crash when performing attended transfer |
ASTERISK-26111: stasis: Inactive Stasis app X missed message |
ASTERISK-26112: res_pjsip: Crash when unlocking dialog in distributor |
ASTERISK-26113: res_pjsip: Lots of DNS lookups of local hostname |
ASTERISK-26114: Testsuite: add support for testing res_odbc |
ASTERISK-26115: pbx: AMI Originate ignore "failed" extension on call failure |
ASTERISK-26116: Crash when performing outgoing call |
ASTERISK-26117: [patch] app_queue: Agent gets auto-paused if SIP client re-registers when call is placed |
ASTERISK-26118: res_srtp: ASAN: heap-use-after-free in ast_srtp_protect |
ASTERISK-26119: [patch] fix: memory leaks, resource leaks, out of bounds and bugs |
ASTERISK-26120: astobj2.c : Asterisk crash, Segmentation fault |
ASTERISK-26121: chan_sip: No voice for voice calling using Android portsip |
ASTERISK-26122: paused agent unpused without reason |
ASTERISK-26123: app_dial saves files under the astdatadir (sounds/priv-callerintros) |
ASTERISK-26124: res_agi: Set audio format for EAGI audio stream |
ASTERISK-26125: sounds: Missing Prompts in Core French Prompt File |
ASTERISK-26126: [patch] leverage 'bindaddr' for TLS in http.conf |
ASTERISK-26127: res_pjsip_session: Crash due to race condition between res_pjsip_session unload and timer |
ASTERISK-26128: Alembic scripts are failing |
ASTERISK-26129: res_rtp_asterisk: Memory leak of CERT bio in DTLS implementation |
ASTERISK-26130: [patch] WebRTC: Should use latest DTLS version. |
ASTERISK-26131: chan_sip: Crash Asterisk (in sip_request_call at chan_sip.c) by making a call to a single character in a dot pattern match |
ASTERISK-26132: PJSIP: provide transport type with received messages |
ASTERISK-26133: app_queue: Queue members receive multiple calls |
ASTERISK-26134: ari: Adding channel to bridge unsubscribes Stasis app from wildcard bridge events |
ASTERISK-26135: ari: Event subscribe and unsubscribe are not symmetric |
ASTERISK-26136: pjsip deadlock after a while when a transport config is wrong |
ASTERISK-26137: PROBLEM WITH TRUNK |
ASTERISK-26138: chan_unistim: Under FreeBSD, chan_unistim generates a compile error |
ASTERISK-26139: test_res_pjsip_scheduler: Compile failure if pjproject isn't installed in a system location |
ASTERISK-26140: res_rtp_asterisk: gcc 6 caught a self-comparison |
ASTERISK-26141: res_fax: fax_v21_session_new leaks reference to v21_details |
ASTERISK-26142: chan_mobile: Audio issues when connected to chan_sip or chan_pjsip |
ASTERISK-26143: res_rtp_asterisk: One way audio when transcoding |
ASTERISK-26144: Crash on loading codecs g729/g723 |
ASTERISK-26145: pjsip: Deadlock with suspend + masquerade + indicate |
ASTERISK-26146: cdr_pgsql: Schema change happened |
ASTERISK-26147: Segfault pj_atomic_dec_and_get () from /lib64/libpj.so.2 |
ASTERISK-26148: pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..." |
ASTERISK-26149: format: Crash when calling from queue |
ASTERISK-26150: Memory leakage in stringfields.c and logger.c |
ASTERISK-26151: pjsip: AOR regex based retrieval does not escape characters |
ASTERISK-26152: Crash issue in ast_format_get_type of format.c |
ASTERISK-26153: manager: Crash when using PresenceState |
ASTERISK-26154: There is no Custom fields on second CDR |
ASTERISK-26155: Error message pops up every few calls, call drop |
ASTERISK-26156: Voice mail envelope |
ASTERISK-26157: Build: Fix errors highlighted by GCC 6.x |
ASTERISK-26158: Asterisk 13.1-cert1 : no RTP when using local channels and sip channel in extensions.conf |
ASTERISK-26159: res_hep: enabled by default and information sent to default address |
ASTERISK-26160: pjsip: Updated->Reachable during qualify |
ASTERISK-26161: chan_sip: Channels remain |
ASTERISK-26162: config: Unable to load config file log level inconsistency. |
ASTERISK-26163: stasis: Documentation lacking on getting ChannelVarset for global variables |
ASTERISK-26164: XMPP no longer triggers NOTIFY to device via chan_pjsip |
ASTERISK-26165: Time of registration |
ASTERISK-26166: res_pjsip_pubsub: Crash when decrementing reference count of message |
ASTERISK-26168: --- SOLVED --- Please close app_mixmonitor stop during a call |
ASTERISK-26169: format_ogg_vorbis: Memory leak using OGG in MixMonitor |
ASTERISK-26170: cdr_odbc: Insertion of record fails with Postgresql |
ASTERISK-26171: REF_DEBUG: support writing storage address to refs log. |
ASTERISK-26172: res_sorcery_realtime: fix bug when successful sql UPDATE is treated as failed if there is no affected rows. |
ASTERISK-26173: func_cdr: CDR function does not permit empty values to be assigned |
ASTERISK-26174: res_pjsip: Crash when freeing cloned message in distributor |
ASTERISK-26175: in CLI, ctrl-W erases entire line like ctrl-U - but should only erase to beginning of current word |
ASTERISK-26176: chan_sip: Add AccountCode to AMI PeerEntry |
ASTERISK-26177: func_odbc: Database handle is kept when it should be released |
ASTERISK-26178: Setting custom SIP headers in PJSIP when using ARI |
ASTERISK-26179: chan_sip: Second T.38 request fails |
ASTERISK-26180: PJSIP: provide valid tcp nodelay option for reuse |
ASTERISK-26181: REF_DEBUG: Node object incorrectly logged during duplicate replacement |
ASTERISK-26182: A second call into inbound queue causes all calls that came in via the queue to go silent. |
ASTERISK-26183: alembic: error when using sqlalchemy version 1.1.0b2 |
ASTERISK-26184: chan_sip: Reference leaks in error paths. |
ASTERISK-26185: Stringfields ABORT |
ASTERISK-26186: res_pjsip_multihomed: Deadlock when finding message header |
ASTERISK-26187: jitterbuf: Crash when putting frame into fixed size buffer |
ASTERISK-26188: asterisk worked fine with one nic eth0 but when i placed it to another server with 2 nics eth0 eth1 i cant make calls |
ASTERISK-26189: Asterisk supports EUROCAE ED137 |
ASTERISK-26190: [patch] SRTP: Enable AES-256 and AES-GCM. |
ASTERISK-26191: threadpool: Leak on duplicate taskprocessor for ast_threadpool_serializer_group |
ASTERISK-26192: ARI: channel hangup make asterisk crash (ast_hangup) |
ASTERISK-26193: chan_sip: reference leak in mwi_event_cb |
ASTERISK-26194: res_pjsip: At startup when using realtime all endpoints are retrieved |
ASTERISK-26195: static analysis: Out of bound array access |
ASTERISK-26196: pbx: Time based includes can leak timezone string |
ASTERISK-26197: chan_sip: Deadlock during hangup |
ASTERISK-26198: ERROR we couldn't allocate a port for RTP instance Revisited |
ASTERISK-26199: PJSIP: tx_data_destroy called twice |
ASTERISK-26200: [patch] res_pjsip_mwi: improve realtime performance - remove unneeded check on endpoint's contacts. |
ASTERISK-26201: Asterisk fails to re-activate an inactive media session (after remote HOLD) |
ASTERISK-26202: core: Random crashes when a channel joins a bridge in formats |
ASTERISK-26203: res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels |
ASTERISK-26204: Locks in chan_sip |
ASTERISK-26205: clang: BlocksRuntime does not link on Fedora 23 |
ASTERISK-26206: [patch] res_pjsip: Use more compatible regex for get all |
ASTERISK-26207: [patch] sRTP: Count a roll-over of the sequence number even on lost packets. |
ASTERISK-26208: pjproject: Use after free when sending packet |
ASTERISK-26209: res_pjsip: Crash when unregistering dialog during subscription reconstruction |
ASTERISK-26210: chan_sip: NOTIFYs for BLF start queuing up and fail to be sent out, Asterisk 13 |
ASTERISK-26211: Unit tests: AST_TEST_DEFINE should be used in conditional code. |
ASTERISK-26212: [patch] Makefile: Retain XML Declaration and DTD in docs. |
ASTERISK-26213: app_meetme: crash at meetme list <confname>: null channel |
ASTERISK-26214: Allow arbitrary time for fax detection to end on a channel |
ASTERISK-26215: app_mixmonitor: No sound when using mixmonitor |
ASTERISK-26216: res_fax: Deadlock when detect fax while channel executing Playback |
ASTERISK-26217: [patch] Codec 2 Mode 2400 |
ASTERISK-26218: [patch] iLBC 20 |
ASTERISK-26219: segfault in ast_manager_build_channel_state_string_prefix |
ASTERISK-26220: Add support for noreturn function attributes. |
ASTERISK-26221: chan_sip: iLBC does not include correct mode |
ASTERISK-26222: res_xmpp: Crash in OpenSSL |
ASTERISK-26223: chan_sip: Changes to Encryption option not accepted on a reload of chan_sip |
ASTERISK-26224: Asterisk not sending out some rtp packets from ConfBridge App |
ASTERISK-26225: Failed to compile |
ASTERISK-26226: pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts |
ASTERISK-26227: sqlalchemy error due to long identifier name |
ASTERISK-26228: res_pjsip_sdp_rtp: G729A does not include annexb=no attribute. |
ASTERISK-26229: [patch] app_voicemail: Add taskprocessor alert level options. |
ASTERISK-26230: [patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup |
ASTERISK-26231: pbx_dundi: Crash if bindaddr is not local |
ASTERISK-26232: Testing 123 |
ASTERISK-26233: pbx: Failure to remove inconsistent extension names |
ASTERISK-26234: chan_sip: CANCEL is not sent to device that responded with 422 Session Interval Too Small to original INVITE |
ASTERISK-26235: res_odbc: Crash when disconnecting |
ASTERISK-26236: core: DSP can be fed slin with different rate than setup. |
ASTERISK-26237: Fax is detected on regular calls. |
ASTERISK-26238: res_pjsip: Empty global default_from_user causes crash |
ASTERISK-26239: res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname |
ASTERISK-26240: chan_sip: No Sound on playback when dialing on Video Call |
ASTERISK-26241: res_pjsip: When using compact headers, rpid and pai are incorrectly generated |
ASTERISK-26242: res_pjsip_endpoint_identifier_dpma crashing with bundled pjsip |
ASTERISK-26243: res_pjsip_session: Direct media not attempted when video codecs enabled |
ASTERISK-26244: contrib/scripts/install_prereq can remove ssh daemon on ubuntu |
ASTERISK-26245: Organize code so utils can use more unmodified Asterisk source files and less ugly hacks. |
ASTERISK-26246: Security: Privilege escalation by AMI adding dialplan extensions. |
ASTERISK-26247: Change system clock lost ip peers |
ASTERISK-26248: chan_pjsip: Error when calling PJSIP client with domain specified |
ASTERISK-26249: app_confbridge: Compliling Asterisk - confbridge module issue |
ASTERISK-26250: Asterisk crashes on unregister with PJSIP multi domain account |
ASTERISK-26251: "400 Bad Request" error on Siemens C470ip SMS via pjsip |
ASTERISK-26252: Segfault when using SendFax / ReceiveFax via T.38 |
ASTERISK-26253: sdp_srtp: libsrtp now a required dependency, shouldn't be |
ASTERISK-26254: Testing 123 |
ASTERISK-26255: Fix includes of sys/signal.h |
ASTERISK-26256: [patch] SIP/SDP origin (o=) contains brackets with IP6 |
ASTERISK-26257: AGI scripts called from dialplan disturb confbrige audio quality |
ASTERISK-26258: SIP response |
ASTERISK-26259: audohook: Crash when duplicating frame |
ASTERISK-26260: Alembic doesn't create extensions. Python error |
ASTERISK-26261: Segfault with libmysqlclient.so |
ASTERISK-26262: stasis: Playing MOH to bridge with ARI does not work |
ASTERISK-26263: SQL error when using realtime and registering extension / inserting into ps_contacts |
ASTERISK-26264: res_pjsip: Crash when applying ACL from non-existent endpoint |
ASTERISK-26265: Errors ignored from some parts of system initialization. |
ASTERISK-26266: SIP Outbound Registration is not working correct |
ASTERISK-26267: ast_register_atexit callbacks should be run on failed startup. |
ASTERISK-26268: alembic: 'auth_username' not in PJSIP 'identify_by' enum |
ASTERISK-26269: res_pjsip: Wrong state for aors without registered contacts after startup |
ASTERISK-26270: Asterisk crashes with double free or corruption on AGI request |
ASTERISK-26271: DTMF(rfc2833 or inband) not working on asterisk 11.22 and DAHDI 2.9.1.1 |
ASTERISK-26272: chan_sip: File descriptors leak (UDP sockets) |
ASTERISK-26273: core: Won't compile when LOW_MEMORY is enabled |
ASTERISK-26274: Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?) |
ASTERISK-26275: Japanese sound file needs to be changed |
ASTERISK-26276: Feature Request Asterisk 13 Pjsip Manipulate To: Header on Dial |
ASTERISK-26277: Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment |
ASTERISK-26278: asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM. |
ASTERISK-26279: pjproject-bundled: Fails to compile on Debian 6 |
ASTERISK-26280: DNS lookups can block channel media paths |
ASTERISK-26281: chan_pjsip would send INVITE to 'Unreachable' endpoints |
ASTERISK-26282: AEL: macro-call in Dial application, macro "lacks 's' extension" |
ASTERISK-26283: res_resolver_unbound: fails configure on older Ubuntu and CentOS |
ASTERISK-26284: pjsip: Deadlock with suspend + masquerade (Alternate scenarios) |
ASTERISK-26285: 'address' audio in all sounds is low quality. |
ASTERISK-26286: ERROR[9264][C-0000072f] astobj2.c: user_data is NULL |
ASTERISK-26287: Asterisk working on create ICE session for RTP instance for unnormal period of time |
ASTERISK-26288: followme: fails to reset config items to default values on reload |
ASTERISK-26289: Announcer channels in ConfBridges cause inefficiencies |
ASTERISK-26290: User login - double free or corruption (out) |
ASTERISK-26291: res_pjsip_session: segfault on already disconnected session |
ASTERISK-26292: app_confbridge: 3D-Conferencing via Binaural Synthesis |
ASTERISK-26293: ami: Connection is reset |
ASTERISK-26294: build: --disable-dev-mode precludes --with-pjproject-bundled in ./configure |
ASTERISK-26295: mISDN/tmp channels and Quality on bridge modules |
ASTERISK-26296: Allow ARI to not require AUTH |
ASTERISK-26297: Delayed SIP signalling and RTP distortion caused by DEBUG_THREADS compile option |
ASTERISK-26298: ari: Swagger resources.json does not acknowledge http prefix as configured in http.conf |
ASTERISK-26299: app_queue: Queue application sometimes stops calling members with Local interface |
ASTERISK-26300: lost supported: path using webRTC |
ASTERISK-26301: rtp_engine: ast_rtp_codecs_payload_code Crash |
ASTERISK-26302: build: Make clean doesn't clean the res/ari directory |
ASTERISK-26303: [patch] BuildSystem: ca_list_path capabilities not detected in PJProject. |
ASTERISK-26304: pjproject 2.5 returns an error if an AAAA lookup returns SERVER FAIL even if the A record lookup worked. |
ASTERISK-26305: Asterisk 14: Two resolver unbound testsuite tests fail |
ASTERISK-26306: channel: Hang-up crashes, chan_pjsip not cleaning up properly |
ASTERISK-26307: res_pjsip_caller_id: Crash on outgoing change |
ASTERISK-26308: compilation failed with -Werror=maybe-uninitialized |
ASTERISK-26309: [patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations. |
ASTERISK-26310: Crash occurs every 24 - 48 hours with backtrace log showing fault related to pjsip hash |
ASTERISK-26311: [patch] rtp_engine: Allow more than 32 dynamic payload types. |
ASTERISK-26312: pbx_lua: Crash occurs when lua code executing |
ASTERISK-26313: chan_sip : Asterisk restart seems to be required for changing encryption option |
ASTERISK-26314: MulticastRTP crashes in 13 as of 13.10 |
ASTERISK-26315: chan_sip does not produce outbound registration - Negative time interval |
ASTERISK-26316: res_pjsip_callerid: Irregular URI causes unexpected callerid |
ASTERISK-26317: res_pjsip_session: Add ability to use preferred codec only |
ASTERISK-26318: Dial with M(macro-name) needs an optional media bridge but don't answer |
ASTERISK-26319: [patch] res_pjsip: qualify/unqualify added/deleted realtime endpoints |
ASTERISK-26320: ari.asterisk.org resets values on page reload |
ASTERISK-26321: ARI : Add reason answered_elsewhere to channel hangup |
ASTERISK-26322: hangup handlers not working from lua |
ASTERISK-26323: hangup handlers not working from lua |
ASTERISK-26324: Change number rings before voicemail |
ASTERISK-26325: Multicast causes Asterisk to crash in 13.10 |
ASTERISK-26326: Crash when dialing MulticastRTP channel |
ASTERISK-26327: res_xmpp.c motif google voice login fails when username domain isnt google |
ASTERISK-26328: Asterisk 14 has changed the output of the AMI Command Call, and it's now unparseable |
ASTERISK-26329: Dialplan: Secure Bridge Media + Secure Bridge Signaling Not Implemented in PJSIP |
ASTERISK-26330: app_queue: Changing the "ringinuse" parameter of a queue doesn't affect dynamic members |
ASTERISK-26331: Crash on “core show channeltype Surrogate” in ast_format_cap_get_names |
ASTERISK-26332: Asterisk 14 won't load g729 |
ASTERISK-26333: Problems with Blind Transfer, PJSIP (Aastra 6869i) |
ASTERISK-26334: res_pjsip_phoneprov_provider.so: undefined symbol: ast_phoneprov_std_variable_lookup |
ASTERISK-26335: function DENOISE() is not present |
ASTERISK-26336: there is a signaling but no voice |
ASTERISK-26337: chan_sip: Received "Forbidden" on qualify messages |
ASTERISK-26338: chan_sip.c:4050 retrans_pkt: Retransmission timeout reached on transmission |
ASTERISK-26339: asterisk change the codec of inbound channel when bridge to another channel. |
ASTERISK-26340: PJSIP 2.5.5 DNS error with IPv4 |
ASTERISK-26341: ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list |
ASTERISK-26342: [patch] Disable -march=native when cross-compiling |
ASTERISK-26343: ASTERISK-25951 causes issues for callerid manipulation through agi |
ASTERISK-26344: Asterisk 13.11.0 + PJSIP crash |
ASTERISK-26345: Asterisk freeze/Crash |
ASTERISK-26346: Asterisk is not listening to both IPv6 and IPv4 |
ASTERISK-26347: Asterisk not sending out RTP packets in case of redirect call |
ASTERISK-26348: chan_sip: File descriptors leak (UDP sockets) also triggered by same-callid |
ASTERISK-26349: 13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed |
ASTERISK-26350: safe_asterisk script fails to start if there is no /dev/tty9 (openvz/lxc) |
ASTERISK-26351: astobj2: Header macro's should deal with error's from ao2_ref. |
ASTERISK-26352: Astcanary dies when doing "core restart" |
ASTERISK-26353: res_musiconhold: musiconhold seems to think that the general section is a class and issues warning |
ASTERISK-26354: Cannot make outgoing calls from SIP account |
ASTERISK-26355: ari: Swagger basePath url always set to http protocol |
ASTERISK-26356: menuselect: invalid test for GTK2 |
ASTERISK-26357: core: AMI binds to random port every Asterisk restart on FreeBSD |
ASTERISK-26358: chan_sip: Contact is updated on re-200, but not on re-INVITE |
ASTERISK-26359: [patch] cdr_mysql: fails to use UTC if so instructed |
ASTERISK-26360: app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs. |
ASTERISK-26361: App VoiceMail msg_id is not unique |
ASTERISK-26362: res_config_mysql: Broken after 13.10 |
ASTERISK-26363: res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code |
ASTERISK-26364: res_pjsip: Don't assume a request will have target addresses |
ASTERISK-26365: rtp: Offer with multiple payloads for same codec is incorrectly handled |
ASTERISK-26366: rtp: RTCP messages with REMB trigger fast picture update |
ASTERISK-26367: rtp: Timestamps broken when video frame is across multiple RTP packets |
ASTERISK-26368: res_pjsip_session: Failover does not update SDP quite right |
ASTERISK-26369: res_pjsip_session: Failover occurs even after channel is hung up |
ASTERISK-26370: bridge_softmix: softmix_bridge_write_voice doesn't handle dsp_talking_threshold correctly |
ASTERISK-26371: Asterisk freezing randomly |
ASTERISK-26372: dial: Crash when removing dial masquerade datastore from peer |
ASTERISK-26373: channel: Crash when appending cap during dialing to a channel, when built with debugging compiler options |
ASTERISK-26374: res_pjsip_multihomed: Contact port is rewritten for connectionful protocols |
ASTERISK-26375: res_pjsip_transport_management: Log message states seconds, but time value is milliseconds |
ASTERISK-26376: res_pjsip_sdp_rtp: Multiple audio streams are not properly discarded |
ASTERISK-26377: res_pjsip: sips URIs are enforced, some agree with this, some don't |
ASTERISK-26378: secure_bridge_signaling and secure_bridge_media always empty |
ASTERISK-26379: res_pjsip_multihomed: SDP rewriting with different IP version |
ASTERISK-26380: res_pjsip_outbound_registration / res_pjsip_outbound_publish: Failover does not occur |
ASTERISK-26381: app_queue: Crash possibly related to very long dial string |
ASTERISK-26382: Double SIP CANCEL with 2 Reason headers |
ASTERISK-26383: Wiki Documentation: document Astcanary |
ASTERISK-26384: Segfault pjsip_endpt_handle_events2 |
ASTERISK-26385: Asterisk with integration Eurocae ED-137b |
ASTERISK-26386: No audio when calling from Asterisk through sip.us and back in |
ASTERISK-26387: Asterisk segfaults shortly after starting even with no active calls. |
ASTERISK-26388: PJSIP stops working, the only clue being the log message "taskprocessor.c: The 'pjsip/distributor-0000XXXX' task processor queue reached 500 scheduled tasks." |
ASTERISK-26389: res_odbc: Clean up pooling options |
ASTERISK-26390: Mixmonitor is stoping to save on file after using AMI Bridge Action |
ASTERISK-26391: Consoles do not display verbose logger messages even when requested. |
ASTERISK-26392: Asterisk reports error in CDR data insertion, but data is available in DB |
ASTERISK-26394: Strange warning messages in JabberSendGroup function |
ASTERISK-26395: IAX2 doesn't support SILK |
ASTERISK-26396: chan_pjsip: HANGUPCAUSE return the wrong code when dialed channel answer. |
ASTERISK-26397: manager: PresenceState action crashes Asterisk 14 |
ASTERISK-26398: core: Remove ABI differences of LOW_MEMORY |
ASTERISK-26399: app_queue: Agent not called when caller is parked |
ASTERISK-26400: app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime |
ASTERISK-26401: Crash on high load |
ASTERISK-26402: Coredump in asterisk 13.6 |
ASTERISK-26403: Asterisk SIP not able to connect from serve r. |
ASTERISK-26404: Add a warning when the same identify IP address exists for 2 endpoints in pjsip.conf |
ASTERISK-26405: alembic: hidefromdir missing from voicemail_users |
ASTERISK-26406: Segurança da Informação - Uma abordagem de segurança utilizando os |
ASTERISK-26407: packages.asterisk.org repos corrupted |
ASTERISK-26408: res_rtp_asterisk: Crash when adding ICE candidate and running under PowerEL7 |
ASTERISK-26409: codec_opus: Update Asterisk to support the translation codec. |
ASTERISK-26410: core: Asterisk 14 doesn't show the header in the console or verbose when starting |
ASTERISK-26411: res_pjsip: Crash pjsip_distributor find_dialog |
ASTERISK-26412: build: Prepare for gcc 6.2 |
ASTERISK-26413: pjproject_bundled: Raise ICE candidate limit from 16 to 32 |
ASTERISK-26414: app_externalivr: ExternalIVR attended transfer - no audio |
ASTERISK-26415: cdr: CDR merged to first call on attended transfer |
ASTERISK-26416: pjproject-bundled: configure fails to check for all required utilities |
ASTERISK-26417: contrib: contrib/scripts/get_mp3_source.sh is empty |
ASTERISK-26418: res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP |
ASTERISK-26419: audiohooks: Remove redundant codec translations when using audiohooks |
ASTERISK-26420: res_ari: Crash when 1 of the channel is leaving a bridge with more participants |
ASTERISK-26421: Segmentation Fault with ARI originate into mixing bridge with 43 clients |
ASTERISK-26422: [patch] Force calendars to do new fetch after module reload |
ASTERISK-26423: res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness |
ASTERISK-26424: app_voicemail: Undocumented behavior from VMSayName |
ASTERISK-26425: download_externals: ignore xmlstarlet return code for optional element |
ASTERISK-26426: format_ogg_opus: remove from source |
ASTERISK-26427: res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip |
ASTERISK-26428: codec_opus: No Audio when using Codec Opus from Chrome WebRTC Web Audio Stream |
ASTERISK-26429: res_xmpp: Custom device state not updated on server it is created on |
ASTERISK-26430: res_hep: will fail, when an IPv6 interface is enabled on the installed server |
ASTERISK-26431: Queues doesn't appear when using realtime configuration |
ASTERISK-26432: chan_dahdi: hangup during long TDD sendtext not detected, blocking channel |
ASTERISK-26433: chan_sip: Allows To-tag checks to be bypassed, setting up new calls |
ASTERISK-26434: Opus binary will not install in Asterisk 13.11.2 |
ASTERISK-26435: Asterisk freezing randomly |
ASTERISK-26436: res_sorcery_realtime: Aliases are filtered |
ASTERISK-26437: Unable to compile on Centos 6.8 when selecting to use pjsip |
ASTERISK-26438: [patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response. |
ASTERISK-26439: chan_rtp: Crash when originating |
ASTERISK-26440: ami: AMI atxfer not accepting alpha numerics |
ASTERISK-26441: crash in ast_rtp_ice_add_cand |
ASTERISK-26442: chan_sip sip debug does not see messages and invites are not processed |
ASTERISK-26443: Asterisk Random Core Dump / Crash |
ASTERISK-26444: 'features show' command in CLI does not return prompt. |
ASTERISK-26445: rtp: Deadlock in getting payload code |
ASTERISK-26446: app_dial: There's no way to override the hangupcause on unanswered channels |
ASTERISK-26447: chan_sip: Better docs needed for SIP_NAT_FORCE_RPORT and SIP_NAT_RPORT_PRESENT |
ASTERISK-26448: Channels in Ringing state after PickUpChan of a local channel |
ASTERISK-26449: Add dockerfile to repo and official dockerhub repo |
ASTERISK-26450: core: Change linked list to other list type |
ASTERISK-26451: TestSuite. Test 15 failed |
ASTERISK-26452: cel_odbc: Memory leak on reading configuration |
ASTERISK-26453: res_pjsip_config_wizard: Memory leak in module_unload |
ASTERISK-26454: CallerID for BLF Not Displaying Number or Name |
ASTERISK-26455: cdr_radius / cel_radius: try fix memory leak |
ASTERISK-26456: format_cap: Crash when running tests/pbx/call_file_retries_alwaysdelete test |
ASTERISK-26457: [patch] force_rport,auto_comedia: No NAT detection triggered. |
ASTERISK-26458: res_pjsip: Insufficient handling of dynamic public IP address |
ASTERISK-26460: audiohooks: Crash when processing interpolated frame |
ASTERISK-26461: chan_sip: SIP MESSAGE body parsed incorrectly (as headers) if first character is a space (0x20) |
ASTERISK-26462: [patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage |
ASTERISK-26463: Asterisk reloads service in the application config |
ASTERISK-26464: Pickup() semantics changed between 1.4 and 1.8 |
ASTERISK-26465: Unable to install asterisk 1.8 on ubuntu 16.04 tls |
ASTERISK-26466: core: Be forgiving on external callerid that may be flawed so we don't drop events |
ASTERISK-26467: chan_pjsip: segfault on hangup, at channel_internal_api.c |
ASTERISK-26468: ari: Bridge events stop working after this sequence of ARI calls |
ASTERISK-26469: Infinite loop after a dual Redirect |
ASTERISK-26470: ARI: Add an 'asterisk_id' field to outgoing events |
ASTERISK-26471: func_strings: Better documentation for REGEX Function, and minor text fixes |
ASTERISK-26472: Asterisk crashes when invalid Dial string used with PJSIP |
ASTERISK-26473: menuselect: GTK frontend messed up on a right-to-left locale |
ASTERISK-26474: res_musiconhold: Crash when scanning files with realtime |
ASTERISK-26475: create dynamic IVR |
ASTERISK-26476: chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings" |
ASTERISK-26477: pjproject: SEGV during SSL operations |
ASTERISK-26478: codec_opus: Opus transcoding one-way audio issue |
ASTERISK-26479: MixMonitor - AUDIOHOOK_INHERIT not functioning on outdoing calls attendant transfers |
ASTERISK-26480: [patch] CLI: core set debug: Auto-completes File not Module |
ASTERISK-26481: FILE function grabs garbage along with read data when target line has no newline |
ASTERISK-26482: [patch] chan_pjsip: segfault on already disconnected session |
ASTERISK-26483: segfault in queued_task_pushed - runs on already destroyed pool |
ASTERISK-26484: res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument. |
ASTERISK-26485: app_dial: Callback Macro for Progress and Ringing events |
ASTERISK-26486: Forwarded call from Digium Phone d40/d70 may crash asterisk |
ASTERISK-26487: res_rtp_asterisk: RTP stream frozen when receiving RTCP packets |
ASTERISK-26488: ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands |
ASTERISK-26489: ARI: Infinite wait area with MOH and periodic announcement |
ASTERISK-26490: res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_" |
ASTERISK-26491: pjsip: PJSIP is confused by erroneous Contact header. chan_sip handles it fine. |
ASTERISK-26492: ARI: Add ability to specify channel variables on websocket events |
ASTERISK-26493: Is REMAINDER behaving in intended way? |
ASTERISK-26494: Inconsistent case for some field names in CEL data, collected via ManagerInterface |
ASTERISK-26495: Lost CDR Uniqueid |
ASTERISK-26496: GROUP_COUNT or GROUP_MATCH_COUNT may report an invalid number of channnels when channels are established nearly simultaneously |
ASTERISK-26497: make install downloads x86_32 variants of external modules on non Intel architectures |
ASTERISK-26498: codec formats: attributes not negotiated properly on 200 OK |
ASTERISK-26499: Locked Here: astobj2_container.c line 333 (internal_ao2_traverse) |
ASTERISK-26500: res_rtp_asterisk: DTMF RFC2833 with timestamp 0 are ignored |
ASTERISK-26501: codec_silk: 404 from url in menuconfig for the README in codec_silk |
ASTERISK-26502: release: Asterisk 11.24.0 patch not complete |
ASTERISK-26503: app_voicemail: Asterisk crashes when MailboxExists is used |
ASTERISK-26504: Multiple Asterisk issues failed to auto-close after they were fixed and a release was made |
ASTERISK-26505: Using res_odbc module for querying a longBLOB column causes a big memory usage |
ASTERISK-26506: [patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c |
ASTERISK-26507: Just a test |
ASTERISK-26508: MGCP call flow when going off-hook |
ASTERISK-26509: A few non-critical deprecation warnings when building on Ubuntu 16.10 |
ASTERISK-26510: pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions |
ASTERISK-26511: pjproject timer_b causes delayed RTP instance destruction |
ASTERISK-26512: app_queue: DAHDI queue member isn't delivered calls after the first |
ASTERISK-26513: tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance |
ASTERISK-26514: Super Awesome Company: Don't specify transport in pjsip.conf |
ASTERISK-26515: rtp_engine: Allocate RTP payloads on a per-session basis |
ASTERISK-26516: pjsip: Memory corruption with possible memory leak. |
ASTERISK-26517: pjsip: Memory corruption with possible memory leak. |
ASTERISK-26518: Error loading module 'res_pjsip.so': undefined symbol: ast_sip_session_register_supplement |
ASTERISK-26519: MemLeak: confbridge/conf_config_parser.c |
ASTERISK-26520: codec_opus: Generated fmtp line has no content |
ASTERISK-26521: MemLeak: res/res_pjsip_session.c |
ASTERISK-26522: pjsip: last ubuntu openssl update broke pjsip build |
ASTERISK-26523: chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression |
ASTERISK-26524: astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled. |
ASTERISK-26525: [UBSAN] bridge_holding.c: member access within misaligned address |
ASTERISK-26526: [UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy |
ASTERISK-26527: Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec |
ASTERISK-26528: [UBSAN] strings.h:signed integer overflow in ast_str_case_hash |
ASTERISK-26529: [UBSAN] codec_adpcm.c: runtime error: left shift of negative value -4 |
ASTERISK-26530: [UBSAN] g722_decode.c: runtime error: left shift of negative value -192 |
ASTERISK-26531: [UBSAN] g722_encode.c: runtime error: left shift of negative value -1 |
ASTERISK-26532: [UBSAN] codec_g726.c:runtime error: left shift of negative value -12 |
ASTERISK-26533: [UBSAN] stdtime/localtime.c: runtime error: left shift of negative value -1 |
ASTERISK-26534: Queue member stuck in state Not is use and in call after channel redirect to another extension. |
ASTERISK-26535: Asterisk won't start after 13.12.1 update |
ASTERISK-26536: Intermittent one-way audio issues |
ASTERISK-26537: AMI: NewConnectedLine event is not documented |
ASTERISK-26538: codec_opus: Add sample to configs/samples/codecs.conf.sample |
ASTERISK-26539: SDP Mangler |
ASTERISK-26540: cdr_radius: use radcli instead of freeradius-client |
ASTERISK-26541: res_pjsip_sdp_rtp: Restrict number of formats to maximum |
ASTERISK-26542: testsuite: tests/apps/statsd tests dependencies and minversion are wrong |
ASTERISK-26543: chan_pjsip doesn't produce RTCP Sender reports as expected |
ASTERISK-26544: res_rtp_asterisk: Delay in DTLS handshake causes audio setup delay |
ASTERISK-26545: Asterisk core dump while sending MWI with chan_sip, (in sip_send_mwi_to_peer at chan_sip.c) |
ASTERISK-26546: mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2' |
ASTERISK-26547: app_confbridge: kick not work for last channel |
ASTERISK-26548: res_pjsip_pubsub: Subscribe to MWI expires early |
ASTERISK-26549: app_dial: When PickupChan() is used some channels may have incorrect device state |
ASTERISK-26550: segfault error 6 in asterisk[400000+2b8000] |
ASTERISK-26551: Voicemail deadlock when under load with ODBC backend |
ASTERISK-26552: chan_sip: chan_pjsip to chan_sip call, CANCEL received on pjsip, CANCEL generated on chan_sip way too late |
ASTERISK-26553: pjsip: Cannot hear transcoded sound files in a g722 call |
ASTERISK-26554: format_h264: ControlPlayback() running slowly |
ASTERISK-26555: Multi-party Video: Fix some post Asterisk-11 regressions |
ASTERISK-26556: manager: AMI version report same in Ast 13 & 14, despite Ast 14 syntax changes |
ASTERISK-26557: Client can't reconnect to a conference with announcements if the client is suddenly killed |
ASTERISK-26558: app_queue: add variable to know if the call is not answered after a queue |
ASTERISK-26559: app_queue: New service level calculation |
ASTERISK-26560: sounds: en_GB core and extra sounds anomalies. |
ASTERISK-26561: chan_unistim on 11, 13, 14, UDP (rtp) ports left unclosed indefinitely |
ASTERISK-26562: app_controlplayback: Transmit Silence on ControlPlayback pause |
ASTERISK-26563: core: macOS devmode build fails: variable 'freeswap' set but not used |
ASTERISK-26564: codec_silk: Very poor sound quality in SiLK implementation |
ASTERISK-26565: chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set |
ASTERISK-26566: res_rtp_asterisk: RTT miscalculation in RTCP |
ASTERISK-26567: PJSIP session supplements are called too late |
ASTERISK-26568: pbx_spool: OUTGOING_RETRY variable |
ASTERISK-26569: ari: Redirect does not work over sip trunk |
ASTERISK-26570: Macro allows an infinite loop of dialplan inclusion resulting in a crash |
ASTERISK-26571: res_pjsip: Resolution incorrect when explicit IPv6 transport configured |
ASTERISK-26572: sorcery: Document mappings and how to mix'n'match |
ASTERISK-26573: Some typos in documentation of chan_sip.c |
ASTERISK-26574: Documentation: Need enhancements to the Asterisk Audio and Video Capabilities section |
ASTERISK-26575: testsuite: Need to check PJSIP functionality when res_srtp is not loaded. |
ASTERISK-26576: Asterisk video SIP recording |
ASTERISK-26577: asterisk -rx "......" show error |
ASTERISK-26578: codec_opus: Instructions to install codec_opus via tarball are incomplete |
ASTERISK-26579: codec_opus: Recursiveness when parsing fmtp line |
ASTERISK-26580: [patch] Error during LDAP modify action when user unregisters |
ASTERISK-26581: codec_opus: Poor quality MOH when using DTX or FEC |
ASTERISK-26582: Asterisk seems to ignore the "n" parameter for "disable console colorization" |
ASTERISK-26583: res_pjsip: When using transports in MariaDB table via ODBC; traffic from Asterisk is not seen on the network. |
ASTERISK-26584: [patch] RTCP feedback for codec modules |
ASTERISK-26585: Asterisk certified 13.8-cert1 & 13.8-cert2 & 13.8-cert3 : freeze on 'sip reload' |
ASTERISK-26586: chan_sip: Segfaults upon reload if client with MWI wasn't registered |
ASTERISK-26587: app_originate: Add option to execute gosub prior to dial |
ASTERISK-26588: Unistim Driver Hang |
ASTERISK-26589: res_pjsip_t38: Deadlock in framehook when writing + negotiating SDP |
ASTERISK-26590: Inconsistent crashes in ast_json_free at json.c |
ASTERISK-26591: res_pjsip: Can't exclude codecs with multiple formats |
ASTERISK-26592: Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage |
ASTERISK-26593: chan_sip: One way audio due to RTP bridging when it shouldn't |
ASTERISK-26594: res_pjsip_pubsub: Cannot persist outside using ps_subscription_persistence table |
ASTERISK-26595: ARI: Add the ability to control the source of video in a multi-party mixing bridge |
ASTERISK-26596: Placing call on hold temporarily locks up set |
ASTERISK-26597: func_pjsip_contact: PJSIP_CONTACT function references MD5ed AOR contact |
ASTERISK-26598: Saynumber is trying to get "and" from "digits/" subfolder |
ASTERISK-26599: res_pjsip: High startup time using local configuration files |
ASTERISK-26600: SIP User to User (UUI) |
ASTERISK-26601: res_odbc: Deadlock when getting connection |
ASTERISK-26602: Testsuite: invalid use of alternate /var/tmp path |
ASTERISK-26603: [patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no |
ASTERISK-26604: chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc. |
ASTERISK-26605: codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded. |
ASTERISK-26606: tcptls: Incorrect OpenSSL function call leads to misleading error report |
ASTERISK-26607: res_ari_playback: allow chararacters URI scheme to be controllable |
ASTERISK-26608: Compile and link failures on OpenBSD |
ASTERISK-26609: chan_sip: Wrong handling of 488 Not Acceptable Here during reINVITE |
ASTERISK-26610: Alembic script for voicemail does not create columns for all options available in voicemail.conf |
ASTERISK-26611: res_rtp_asterisk: Fix byte order on non-8kHz signed linear |
ASTERISK-26612: assert failure |
ASTERISK-26613: format_wav: wav16 format read file only by 320 - half of frame |
ASTERISK-26614: app_queue: updatecdr option in queues.conf does effectively nothing |
ASTERISK-26615: Create a version of codec_opus that is not statically linked with libopus |
ASTERISK-26616: chan_sip: T38 calls not being hung up correctly |
ASTERISK-26617: res_rtp_asterisk: Can't bind on systems without IPv6 |
ASTERISK-26618: build: Backport addition of librt check to configure.ac |
ASTERISK-26619: Both AST_ARGS and ASTARGS used in /etc/init.d script |
ASTERISK-26621: app_queue: Queue application does not ring members with Local interface |
ASTERISK-26622: Call ID in AMI Cdr events |
ASTERISK-26623: res_pjsip: Crash when calling PJSIPShowEndpoint |
ASTERISK-26624: res_calendar_caldav: Add support for gmail |
ASTERISK-26625: All PJSIP calls rejected with 'Unable to allocate RTP socket: Address family not supported by protocol' |
ASTERISK-26626: sounds: Submission of en_NZ Sound Prompts |
ASTERISK-26627: func_global: Provide better documentation for usage of SHARED |
ASTERISK-26628: tests/apps/confbridge/confbridge_recording: Failing due to frame deferral changes |
ASTERISK-26629: tests/manager: 4 test failures as a result of iostream change |
ASTERISK-26630: Make logging PJPROJECT messages a bit easier |
ASTERISK-26631: chan_skinny.c: Skinny Client sent less data than expected. Expected 4 but got 0 |
ASTERISK-26632: core: Possibility of a frame "imbalance" leading to stuck channels. |
ASTERISK-26633: res_odbc: SQL Execute returned "Deadlock found when trying to get lock; try restarting transaction" |
ASTERISK-26634: OpenBSD fails to link libasteriskpj.o |
ASTERISK-26635: basic-pbx: pbx_functions.so in modules.conf of basic-pbx sample |
ASTERISK-26636: pjsip stopped workinh on lastest update |
ASTERISK-26637: chan_sip: Video TLS SRTP Broken |
ASTERISK-26638: res_clioriginate: originate Local always creates channel with slin 8khz format |
ASTERISK-26639: core: Disabling xmldoc support does not work. Also results in abort during Asterisk startup. |
ASTERISK-26640: app_queue: Asterisk crashes when dialplan execution goes to queue application |
ASTERISK-26641: Backport PJSIP moh_passthrough to Asterisk 13 |
ASTERISK-26642: res_pjsip_session: Improper Handling of extra 200 OK from forked INVITE |
ASTERISK-26643: Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk |
ASTERISK-26644: PJSIPShowRegistrationsInbound just dumps all aors |
ASTERISK-26645: res_pjsip_endpoint_identifier_ip: Does not check for updates in realtime database |
ASTERISK-26646: CoreShowChannel seems to intermittently get the same Uniqueid Linkedid |
ASTERISK-26647: Support older DNS style for OpenBSD |
ASTERISK-26648: ulimit setting is not working with asterisk |
ASTERISK-26649: Asterisk voicemail with exchange 2010 frequently crashing exchange imap service |
ASTERISK-26650: Asterisk 13.12.0 Crash |
ASTERISK-26651: Using static analyze in Asterisk |
ASTERISK-26652: Asterisk hung channels |
ASTERISK-26653: pjproject_bundled doesn't verify already downloaded tarballs |
ASTERISK-26654: chan_sip: ILBC or Opus codec offer correlates with one-way audio |
ASTERISK-26655: [patch]pjsip: Transfers Broken with Compact Headers Enabled |
ASTERISK-26656: PJSIP fails to handle NAT clients on multihomed boxes |
ASTERISK-26657: Chan_sip deadlock with local channel and audiohooks |
ASTERISK-26658: Add ability for dialplan show to display filenames/line numbers of registered extensions |
ASTERISK-26659: res_pjsip: PJSIP presence - missing braces around the status element in XML |
ASTERISK-26660: SIP trunk Registration error 403 forbidden |
ASTERISK-26661: res_ari: channels/play REST api call should send an error when media resource not found |
ASTERISK-26662: Asterisk crashed in after_bridge callback on normal Dial() call |
ASTERISK-26663: Chan_sip deadlock with local channel and audiohooks |
ASTERISK-26664: pjsip: pj_thread_register() assertion |
ASTERISK-26665: app_queue: Agent ringing, Caller hangup before timeout, no agent name logged - missing RINGNOANSWER? |
ASTERISK-26666: chan_pjsip: Opus offered and used, but uLaw is sent, Result No Audio |
ASTERISK-26667: sorcery: Segfault in res_sorcery_memory_cache, in ast_sorcery_object_get_id at sorcery.c |
ASTERISK-26668: core: Malformed pattern matching extension (various factors) results in crash |
ASTERISK-26669: PJSIP Segfault 13.13.1 (Bundled PJSIP) |
ASTERISK-26670: [patch] Outgoing SIP-URI Dialing via PJSIP |
ASTERISK-26671: app_queue: QueueMemberStatus event missing |
ASTERISK-26672: Crash when setting remote address on RTP instance |
ASTERISK-26673: chan_pjsip: Crash when using CHANNEL dialplan function around masquerade |
ASTERISK-26674: AMI action Park is reported unknown |
ASTERISK-26675: PJSIP Segmentation Fault grp_lock_acquire |
ASTERISK-26676: enabling TEST_FRAMEWORK changes AST_BUILDOPT_SUM |
ASTERISK-26677: app_sms: Short SMS messages are garbled |
ASTERISK-26678: chan_sip: AMI Events - PeerStatus does not show time parameter |
ASTERISK-26679: Crash on invalid contact domain (pjsip aor) |
ASTERISK-26680: How to configure asterisk as a media server |
ASTERISK-26681: chan_local: Local Channels Not Optimizing |
ASTERISK-26682: Device State stay stuck - maybe deadlock |
ASTERISK-26683: res_calendar: Calendars duplicated after module reload |
ASTERISK-26684: res_pjsip: Various issues with compact SIP headers |
ASTERISK-26685: res_pjsip: Crash when using IPv6 and Transport ws,wss |
ASTERISK-26686: res_pjsip: Lock inversion in transport management |
ASTERISK-26687: log_member_name_as_agent is missed in 13.8-cert4 |
ASTERISK-26688: Documentation: voicemail.conf.sample shows 512 limit for emailbody field, however this is only true if compiled with LOW_MEMORY option |
ASTERISK-26689: res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting channel for lack of RTP activity |
ASTERISK-26690: res_pjsip: segfault in ssl_write from pjsip_endpt_process_rx_data |
ASTERISK-26691: Remember SDP negotiation on SIP_CODEC_INBOUND. |
ASTERISK-26692: res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip) |
ASTERISK-26693: res_pjsip_endpoint_identifier_ip: Add support for SRV |
ASTERISK-26694: PJSIP: Notify event not authenticating to the phone. |
ASTERISK-26695: Poor playback quality when using .mp3 files with Playback function |
ASTERISK-26696: pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh |
ASTERISK-26697: Wiki Documentation: Create page that points to or includes resources on the topic of Asterisk & Kamailio integration |
ASTERISK-26698: PJSIP: PAI showing dial plan filters to customers |
ASTERISK-26699: res_pjsip: Assertion when sending OPTIONS request to endpoint |
ASTERISK-26700: pjsip: Crash on startup on WSS client SIP OPTIONS message |
ASTERISK-26701: res_pjsip_sdp_rtp: Optimistic Encryption Doesn't Recognize DTLS fingerprint |
ASTERISK-26702: AMI/PJSIP: ExtensionStatus event incorrectly shows Unavailable |
ASTERISK-26703: PJSIP Opus jitter rating incorrect over AMI RTCP events |
ASTERISK-26704: res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'. |
ASTERISK-26705: libasteriskssl.so not found when asterisk is installed for the 1st time |
ASTERISK-26706: Segfault in dial_target_free stasis_channels.c:1349 |
ASTERISK-26707: Segfault ast_json_free (p=0x7fb100000002) at json.c:190 |
ASTERISK-26708: res_pjsip: ARI interface does not correctly indicate de-registrations |
ASTERISK-26709: res_ari: ARI Snoop-whisper fails (no transmit audio ) to channel in holding bridge |
ASTERISK-26710: [patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0 |
ASTERISK-26711: func_enum: ENUM code wrong case |
ASTERISK-26712: rtp: Continuously increase in memory of asterisk process and causing crash in the process. |
ASTERISK-26713: Segfault when removing object from cache |
ASTERISK-26714: Phone default have not ringing on ARM |
ASTERISK-26715: app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel |
ASTERISK-26716: ari: Channels with pre-dial handlers cannot be hung up via ARI |
ASTERISK-26717: Document the fact that Asterisk HEP support only works with the PJSIP channel driver |
ASTERISK-26718: ARI: Bridge destroying doesn't work as expected |
ASTERISK-26719: pbx: Only up to 127 includes in a dialplan context (AST_PBX_MAX_STACK - 1) |
ASTERISK-26721: Asterisk Crash - astobj2.c: FRACK!, Failed assertion bad magic number 0x0 for object 0x7fc9f402d088 |
ASTERISK-26722: Dialing an early bridged channel by ARI with timeout not equal zero causes bridge and thread leaks |
ASTERISK-26723: VoiceMailPlayMsg not playing messages via realtime |
ASTERISK-26724: Hints appear to not expire properly resulting in duplicates |
ASTERISK-26725: astobj2.c: FRACK!, Failed assertion bad magic number |
ASTERISK-26726: Call app.Dial with "d" option from lua dialplan freezes asterisk until kill the process of asterisk |
ASTERISK-26727: FRACK!, Failed assertion bad magic number 0x0 |
ASTERISK-26728: Unique channel names per server |
ASTERISK-26729: res_pjsip_sdp_rtp: Keepalive does not work on video |
ASTERISK-26730: 13.13.1 with bundled pjproject with optimization used cpu twice more then 13.7.2 |
ASTERISK-26731: res_sorcery_memory_cache: memory leak on every sorcery memory cache populate |
ASTERISK-26732: res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome |
ASTERISK-26733: asterisk queue segfault |
ASTERISK-26734: asterisk-dahdi fails on Fedora 25 due to missing dahdi-tools package |
ASTERISK-26735: res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect |
ASTERISK-26736: PJSIP: one way audio |
ASTERISK-26737: Problema de latencia em alguns ramais da empresa |
ASTERISK-26738: Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c |
ASTERISK-26739: voicemail API test: confuses expected and actual values |
ASTERISK-26740: voicemail API test: uses varlibdir instead of datadir for a sound file |
ASTERISK-26741: PJSIP device state exchange does not work between Asterisk instances on different network subnets |
ASTERISK-26742: Automatically Delete Old Voice Messages |
ASTERISK-26743: PJPROJECT: Detecting compiled max log level does not work. |
ASTERISK-26744: Asterisk fails to start up when using OpenSSL 1.1.0 |
ASTERISK-26745: Asymmetric codecs when asymmetric_rtp_codec=no |
ASTERISK-26746: pjsip: Repeated segfaults with not responding nameserver |
ASTERISK-26747: Ambiguity in PJSIP use_avpf and force_avp |
ASTERISK-26748: app_confbridge: max_members does not limit wait_marked users when they've joined before the marked user |
ASTERISK-26749: chan_dahdi.c: my_dahdi_write() returns zero even if it fails. |
ASTERISK-26750: Applications: XML - New application to parse XML Strings. |
ASTERISK-26751: Applications: JSON - New app to parse JSON into dialplan variables |
ASTERISK-26752: Asterisk 13.8.2 crashes on Debian |
ASTERISK-26753: AMI disconnect causes "ast_careful_fwrite: fwrite() returned error: Broken pipe" |
ASTERISK-26754: build_tools: make_build_h does not handle \ in user name |
ASTERISK-26755: app_queue: Random queues disappear on "core reload queue all" |
ASTERISK-26756: res_pjsip_mwi: Asterisk does not terminate MWI subscription |
ASTERISK-26757: When a queue member transfers queue call, he remains marked as "in call" |
ASTERISK-26758: res_hep_pjsip: For WebRTC clients Asterisk insert same ip address in "source ip address" and "destination ip address" fields in HEP packets |
ASTERISK-26759: Double free, ast_taskprocessor_execute->tps_task_free |
ASTERISK-26760: Statically assigning Hotdesking agent to queue in queue.conf |
ASTERISK-26761: SIGSEGV when user joins confbridge with speex and dtx is active |
ASTERISK-26763: When use FLASH on a FXS DAHDI channel, it doesn't transfer |
ASTERISK-26764: chan_pjsip: Crash looking up PJSIP call-id on hungup channel. |
ASTERISK-26765: res_resolver_unbound: FRACK! Excessive ref count trap tripped. |
ASTERISK-26766: Asterisk 13 IAX2 jitterbuffer causes packet loss |
ASTERISK-26767: ARI channelvars cause memory leak |
ASTERISK-26768: chan_sip: Crashes with more than 90 SIP endpoints with TCP - in __ast_string_field_ptr_grow at stringfields.c, in parse_register_contact at chan_sip.c |
ASTERISK-26769: T38 informations during session progress are not passed to the client |
ASTERISK-26770: res_stasis_device_state: Duplicate subscriptions when multiple received at same time |
ASTERISK-26771: ARI: HTTP requests take 200ms to be taken into account |
ASTERISK-26772: Crash in srv.c on startup with pjsip |
ASTERISK-26773: stream: Add basic API |
ASTERISK-26774: core: Playback URL fails after some time |
ASTERISK-26775: app_queue: reset abandoned in service level |
ASTERISK-26776: res_pjsip_pubsub: Crash when generating xpidf content |
ASTERISK-26777: res_sorcery_memory_cache deadlocks |
ASTERISK-26778: infrastructure: Unit test execution does not have log file |
ASTERISK-26779: chan_pjsip: PJSIP_HEADER does not understand compact headers |
ASTERISK-26780: res_pjsip: PJSIP Registration Fails when transport=transport-udp6 |
ASTERISK-26781: bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio |
ASTERISK-26782: res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication |
ASTERISK-26783: res_pjsip_outbound_registration: line= state not modified on reload |
ASTERISK-26784: res_pjsip_session: Problems with Reinvites when one Endpoint IPv4 and other IPv6 |
ASTERISK-26785: configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample |
ASTERISK-26786: Implement ast_stream_topology API |
ASTERISK-26787: Received incoming SIP connection from unknown peer" from registered sip provider |
ASTERISK-26788: core: Protect flags during ast_waitfor |
ASTERISK-26789: Audit manipulation of channel flags without locks |
ASTERISK-26790: Implement stream topology (non-change request) API usage in channels |
ASTERISK-26791: 14.3.0 download archive corrupt - cannot extract |
ASTERISK-26792: Lock inversion between channel hangup and the bridging code |
ASTERISK-26793: Implement ast_write_stream in channels |
ASTERISK-26794: http: Crash on Reload Only in ast_tcptls_server_start |
ASTERISK-26795: pjproject MD5SUM link incorrect |
ASTERISK-26796: res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS' |
ASTERISK-26797: res_pjsip: Crash when freeing pool of cloned message |
ASTERISK-26798: PJSIP: Limit of 6000 for max packet size is insufficient for some WebRTC INVITEs |
ASTERISK-26799: res_pjsip: Using an auth object for inbound and outbound authentication fails. |
ASTERISK-26800: Non-threadsafe function usage analysis |
ASTERISK-26801: Not able to install Asterisk 10.0.0 on Ubuntu 14.0.4 |
ASTERISK-26802: [patch] Integrity Check Of PJSIP Download Fails |
ASTERISK-26803: res_pjsip: AOR Bindings Erased on REGISTER-Fetch (Endpoint Unreachable) |
ASTERISK-26804: chan_sip: Channel not hung up properly after leaving ConfBridge |
ASTERISK-26805: ast_audiohook_detach_list (audiohook_list=0xdeaddeaddeaddead) & Asterisk crash after participants end calls in confbridge conference |
ASTERISK-26806: pjsip_options: rework to make more efficient |
ASTERISK-26807: sounds: New 3-D Binaural audio features require new sound prompts |
ASTERISK-26808: res_pjsip_outbound_registration doesn't know about network change events |
ASTERISK-26810: codec_opus: Crash when translating frame with opus section |
ASTERISK-26811: stream: Add streams to "core show channel" |
ASTERISK-26812: [patch] Fix download_externals To Allow The Use Of curl Or wget |
ASTERISK-26813: asterisk version 12 crash |
ASTERISK-26814: pjproject_bundled build fails to download pjproject source when using cURL |
ASTERISK-26815: Call recording drops when transferring call to queue |
ASTERISK-26816: Implement ast_read_stream in channels |
ASTERISK-26817: autodestruct on dialog in place (method bye). rescheduling destruction for 10000 ms |
ASTERISK-26818: cdr: Problem setting variables in h exten |
ASTERISK-26819: Crash in xpidf_to_string |
ASTERISK-26820: local channel doesn't respect default language setting in asterisk.conf |
ASTERISK-26821: Subscriptions with Expiry=0 is not removed |
ASTERISK-26822: pjsip/cli_commands: pjsip show channelstats shows wrong codec |
ASTERISK-26823: PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist |
ASTERISK-26824: res_pjsip_transport_websocket: IPv6 does not work |
ASTERISK-26825: pjsip.conf.sample: user_agent: still refers to branch 12 |
ASTERISK-26826: testsuite: Add support for Python 3 |
ASTERISK-26827: asterisk service startup error message |
ASTERISK-26828: Unable to delete object from sorcery cache |
ASTERISK-26829: res_pjsip: Assertion when applying transport configuration via push |
ASTERISK-26830: SEGV in stasis_message_router_publish_sync |
ASTERISK-26831: res_rtp_asterisk: Race condition when RTCP and WebRTC is used |
ASTERISK-26832: res_pjsip: Segfault when calling pjsip_hdr_print_on in sip_msg.c:581 |
ASTERISK-26833: CLONE - app_confbridge: kick not working just after channel entered the conference |
ASTERISK-26834: app_voicemail: UTF-8 string does not get properly used in email |
ASTERISK-26835: res_rtp_asterisk: Crash when freeing RTCP address string |
ASTERISK-26836: res_pjsip: Memory corruption of endpoint |
ASTERISK-26837: res_pjsip: FRACK when qualifying removed endpoint |
ASTERISK-26838: app_queue: Agents getting calls when busy |
ASTERISK-26839: core: Implement stream topology changing in channels |
ASTERISK-26840: Asterisk + WebRTC crash |
ASTERISK-26841: chan_sip: Call not cancelled after receiving a 422 response |
ASTERISK-26842: Websocket becomes disconnected when trying to place call from browser |
ASTERISK-26843: app_directed_pickup: Not working for ring group |
ASTERISK-26844: Wiki documentation: Deployment - Guides for Asterisk configuration with NAT and firewalls - develop documentation |
ASTERISK-26845: Any plans to implement g729 |
ASTERISK-26846: chan_sip: Add rtcp-mux support |
ASTERISK-26847: cdr: Crash when finding CDR record |
ASTERISK-26848: Problem with call return when atxfer |
ASTERISK-26849: Newchannel event doesn't inherit variables |
ASTERISK-26850: res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets |
ASTERISK-26851: res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport |
ASTERISK-26852: res_sorcery_config: Crash when allocating |
ASTERISK-26853: res_rtp_asterisk: Crash in pjnath when receiving packet |
ASTERISK-26854: build: Error while running configure on Solaris |
ASTERISK-26855: cant able to transmit the voice on two different network |
ASTERISK-26856: build: Compile fail with Module Embedding enabled |
ASTERISK-26857: chan_pjsip: Dialplan function race condition |
ASTERISK-26858: chan_sip: SDP does not contain declined video stream in session timers |
ASTERISK-26859: res_odbc: Deadlock when getting connection |
ASTERISK-26860: Upon RTCP reception, netsock2.c:210 ast_sockaddr_split_hostport: Port missing in (null) |
ASTERISK-26861: cdr: Deadlock when batch is unscheduled |
ASTERISK-26862: app_queue: Queue stops calling members with local interface after forwarding in previous call |
ASTERISK-26863: res_pjsip: Add endpoint identification scheme based on a configured SIP header/value |
ASTERISK-26864: res_pjsip_session: Add support for overlap dialling |
ASTERISK-26865: chan_iax2: Reload of iax peer results in loss of host address/port |
ASTERISK-26866: Two devices can't register at once |
ASTERISK-26867: autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade). |
ASTERISK-26868: ARI: Asterisk crash - frame copy into invalid memory during bridging operations |
ASTERISK-26869: res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension |
ASTERISK-26870: codec_opus: Codec configured with constant bit rate, but frame sizes changes |
ASTERISK-26871: Record command don't record video on SIP/PJSIP with VP8/h263p |
ASTERISK-26872: Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal) |
ASTERISK-26873: realtime_odbc: heap-buffer-overflow in SQLGetData |
ASTERISK-26874: chan_sip: SIP_CODEC Without Early Media Causes Odd Transcoding |
ASTERISK-26875: app_mixmonitor: Recording out of sync when 183 but no RTP |
ASTERISK-26876: Multiple registeration |
ASTERISK-26877: app_queue: Crash when seeing if a member can be rung |
ASTERISK-26878: func_channel: Add ability to get the callid so dialplan has access to it. |
ASTERISK-26879: PJSIP external_media_address ignored if no local_net options are provided |
ASTERISK-26880: Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled |
ASTERISK-26881: bridge_channel.c:989 ast_bridge_channel_queue_frame: We couldn't write alert pipe for Local/XXXXXXX@dialpeer. something is VERY wrong |
ASTERISK-26882: Asterisk Memory Curruption |
ASTERISK-26883: pjsip DNS SRV: missing auto generated identify match entry if DNS temporarily fails on startup |
ASTERISK-26884: crash |
ASTERISK-26885: channel: Support dynamic number of file descriptors |
ASTERISK-26886: chan_pjsip: make PJSIP sample file point to wiki documentation a bit more explicitly |
ASTERISK-26887: Segfault pjmedia_sdp_neg_negotiate |
ASTERISK-26888: SIP TLS version |
ASTERISK-26889: Using SIP TLS on port alternate to 5061. |
ASTERISK-26890: STUN server with non-default-route transport causes INVITE delay |
ASTERISK-26891: Re-enable plain WebSockets in pjsip for reverse proxies |
ASTERISK-26892: pbx_dundi: Asterisk segfaults when run in Kickstart, in destroy_peer at pbx_dundi.c |
ASTERISK-26893: No "alert" or "progress" in chan_ooh323 if debug is enabled only on the module |
ASTERISK-26894: pjsip should support tel uri scheme |
ASTERISK-26895: AMI Redirect'd channel hangs up if created from AMI Originate |
ASTERISK-26896: Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT |
ASTERISK-26897: chan_sip: Security vulnerability with client code header |
ASTERISK-26898: PJSIP SDP Packet Size hard coded limit too small for WebRTC |
ASTERISK-26899: Unable to apply outbound proxy on request to qualify |
ASTERISK-26900: sdp: Add support for connection address management and topology updating |
ASTERISK-26901: OPUS doesn't generate SDP accordingly to codecs.conf |
ASTERISK-26902: chan_sip: Unnecessary changes to Domain Part of Request URI when setting User part |
ASTERISK-26903: Listening TCP/TLS sockets stop when temporarily out of open files |
ASTERISK-26904: codec silk crash asterisk on outgoing call |
ASTERISK-26905: pjproject_bundled: Merge 3 upstream deadlock patches into bundled |
ASTERISK-26906: chan_sip: With "callbackextension" set, the "port" option is not respected. |
ASTERISK-26907: Voicemail discarding messages |
ASTERISK-26908: res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked. |
ASTERISK-26909: Libdir defaults to /usr/lib64 on systems that have a stray lib64 directory |
ASTERISK-26910: chan_sip: Intermittently Asterisk doesn't send CANCEL/BYE when making parallel DIAL() |
ASTERISK-26911: chan_sip: Should test case-insensitive for occurence of "RTP/AVP" in SDP RTP profile |
ASTERISK-26912: Asterisk Crash on pj_ice_sess_on_rx_pkt |
ASTERISK-26913: Music on hold restart when second user comes |
ASTERISK-26914: Asterisk freezes for a few seconds under certain queue scenarios |
ASTERISK-26915: chan_sip: Session Timers required but refused wrongly. |
ASTERISK-26916: res_pjsip: Excessive refcount reached on transport ao2 object |
ASTERISK-26918: Failed Subscriptions with Expiry=0 Added and not Removed |
ASTERISK-26919: res_pjsip_dialog_info_body_generator: Ringing&&InUse behavior difference between chan_sip and res_pjsip |
ASTERISK-26920: app_queue: PAUSEALL/UNPAUSEALL does not log reason |
ASTERISK-26921: Restarting service |
ASTERISK-26922: chan_sip: tcpbind uses wrong source address |
ASTERISK-26923: bridging: T.38 request is lost when channels are added to bridge |
ASTERISK-26924: Codec OPUS bad quality |
ASTERISK-26925: asterisk realtime(mysql) : set call duration before call starts |
ASTERISK-26926: func_speex: Crash caused by frame with no datalen |
ASTERISK-26927: pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete(). |
ASTERISK-26928: pjsip: Add database tables for PUBLISH support |
ASTERISK-26929: pjsip: Add database tables for RLS |
ASTERISK-26930: pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux |
ASTERISK-26931: FRACK!, Failed assertion bad magic number framehook_detach() |
ASTERISK-26932: [patch] SIP/SDP: No rtpmap for static RTP payload IDs |
ASTERISK-26933: RTP instance does not use same IP as explicit transport (unspecified transport) |
ASTERISK-26934: res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport (unspecified transport) |
ASTERISK-26935: Asterisk crashes in ast_channel_name at channel_internal_api.c after "FRACK!, Failed assertion bad magic number" |
ASTERISK-26936: how to initialize rtp engine ice? |
ASTERISK-26937: rtp ice is never initialized? |
ASTERISK-26938: Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP |
ASTERISK-26939: Out of bound memory access in PJSIP multipart parser crashes Asterisk |
ASTERISK-26940: Asterisk Skinny memory exhaustion vulnerability leads to DoS |
ASTERISK-26941: ARI WebSocket forcibly closed due to fatal write error on repeated mute/unmute requests |
ASTERISK-26942: pjproject_bundled: Pass asterisk library overrides to pjproject configure |
ASTERISK-26943: Problem with setting accountcode to iax2 friend/peer/user |
ASTERISK-26944: Asterisk issue tracker cleanup |
ASTERISK-26945: Integrating Zoho CRM with Asterisk |
ASTERISK-26946: Crash as the user leaves the conference when the video frame is distributed |
ASTERISK-26947: need ignoresdpversion=1 for PJSIP |
ASTERISK-26948: Fax does not work using T.38 passthrough |
ASTERISK-26949: sdp: Implement T.38 |
ASTERISK-26950: DUNDi errors when instructed to bind to an IPv6 address |
ASTERISK-26951: chan_sip: ACK with SDP does not update a direct media bridge |
ASTERISK-26952: Need CDR reports about the Calls which are put on Hold |
ASTERISK-26953: Asterisk crash if hep.conf have some missing parameters |
ASTERISK-26954: Queue agi variable not returning |
ASTERISK-26955: pjsip: SIP Packets with Via "received=" Containing IPv6 Address Delimited by "[]" Rejected |
ASTERISK-26956: Exceptionally long voice > queue length queuing |
ASTERISK-26957: manipulating ToHeader number with CallerID(DNID) |
ASTERISK-26958: Segfault ast_format_cmp |
ASTERISK-26959: dial: Allow topology of dialing channel to influence dialed channel |
ASTERISK-26960: SIGABRT in snmp/agent.c |
ASTERISK-26961: [UBSAN] chan_sip.c: left shift of 1 by 31 places cannot be represented in type 'int' |
ASTERISK-26962: [UBSAN] asterisk.c: left shift of 1 by 63 places cannot be represented in type 'long int' |
ASTERISK-26963: Crash in ast_manager_build_bridge_state_string_prefix |
ASTERISK-26964: res_pjsip_session: Wrong From on reinvite when request and To URI differ |
ASTERISK-26965: Caller ID set to 'asterisk' during atxfer from bridged queue |
ASTERISK-26966: bridge_simple: Add support for streams |
ASTERISK-26967: Billsec is not as expected- Asterisk 13 |
ASTERISK-26968: chan_pjsip: Transfer() does not result in TRANSFERSTATUS reflecting SIP response to transfer |
ASTERISK-26969: Deadlock - chan_pjsip |
ASTERISK-26970: Core crash with Segmentation fault 0xdead |
ASTERISK-26971: Not Able to Build Asterisk |
ASTERISK-26972: jitterbuffer: Crash in adaptive implementation with large timer interval |
ASTERISK-26973: bridge: Crash when freeing frame and snooping |
ASTERISK-26974: res_pjsip: Deadlock in T.38 framehook |
ASTERISK-26975: app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call |
ASTERISK-26976: libsrtp-2.x.x support |
ASTERISK-26977: Unable to install on Ubuntu |
ASTERISK-26978: rtp: Crash in ast_rtp_codecs_payload_code() |
ASTERISK-26979: res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110 |
ASTERISK-26980: pjsip: Clean up WebRTC disables |
ASTERISK-26981: contrib: Patch for install_prereq script to support correct packages for Fedora F25 |
ASTERISK-26982: chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable |
ASTERISK-26983: Crash in Manager Reload when TLS Config Changes |
ASTERISK-26984: chan_pjsip TLS incorrectly tears down connection |
ASTERISK-26985: file: cache_record_files does not respect append flag |
ASTERISK-26986: Poor high load perfomance, locks,lost rtp |
ASTERISK-26987: pbx_dundi: Asterisk crashes when unloading module pbx_dundi.so with dundi peers |
ASTERISK-26988: res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs |
ASTERISK-26989: Queue won't call members on calls from THAT queue |
ASTERISK-26990: Asterisk 11.25.1 and 11.24.1 crash |
ASTERISK-26991: documentation: Doxygen site is no longer being updated |
ASTERISK-26992: chan_sip: Bell Canada Interop on Asterisk 13.15 fails due to no RTP on SDP sendrecv |
ASTERISK-26993: app_chanspy: ExtenSpy on extension with no active channels |
ASTERISK-26994: Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name |
ASTERISK-26995: Add QUEUE_FLOAT_PENALTY to app_queue |
ASTERISK-26996: chan_pjsip: Flipping between codecs |
ASTERISK-26997: Create an StreamEcho dialplan application |
ASTERISK-26998: res_pjsip_session: INVITE retransmissions could still setup the same call again. |
ASTERISK-26999: Asterisk crashes while starting, Inconsistency detected by ld.so |